Hi, I have done a source installation of gstreamer and gstreamer-base plugins version 0.10.21 I am trying few things mentioned in the faq document If I say $aplay -v test.wav, I am able to play the wav file through alsaplay, which shows that alsa driver is installed in my PC If I do $gst-inspect alsasink shows that alsasink is installed. And when i do $gst-launch filesrc location=test.ogg ! decodebin ! audioconvert ! audioresample ! filesink location=test.raw it writes the decoded raw samples into the file and I am able to play test.raw in application like cooledit But if I do $gst-launch filesrc location=test.ogg ! decodebin ! audioconvert ! audioresample ! alsasink it goes into infinite loop. The alsasink plugin is not able to detect the alsadriver I guess. How do I debug this problem? Pls help me Regards Vinod James ------------------------------------------------------------------------------ SF.Net email is Sponsored by MIX09, March 18-20, 2009 in Las Vegas, Nevada. The future of the web can't happen without you. Join us at MIX09 to help pave the way to the Next Web now. Learn more and register at http://ad.doubleclick.net/clk;208669438;13503038;i?http://2009.visitmix.com/ _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
On Fri, Dec 5, 2008 at 8:30 AM, vinod james <[hidden email]> wrote:
> > Hi, > I have done a source installation of gstreamer and gstreamer-base plugins > version 0.10.21 > I am trying few things mentioned in the faq document > If I say > $aplay -v test.wav, > I am able to play the wav file through alsaplay, which shows that alsa > driver is installed in my PC > > If I do > $gst-inspect alsasink > shows that alsasink is installed. > > And when i do > $gst-launch filesrc location=test.ogg ! decodebin ! audioconvert ! > audioresample ! filesink location=test.raw > it writes the decoded raw samples into the file and I am able to play > test.raw in application like cooledit > > But if I do > $gst-launch filesrc location=test.ogg ! decodebin ! audioconvert ! > audioresample ! alsasink > it goes into infinite loop. > The alsasink plugin is not able to detect the alsadriver I guess. > How do I debug this problem? > Pls help me Try with: gst-launch-0.10 audiotestsrc ! alsasink And for debugging: export GST_DEBUG=alsa:5 -- Felipe Contreras ------------------------------------------------------------------------------ SF.Net email is Sponsored by MIX09, March 18-20, 2009 in Las Vegas, Nevada. The future of the web can't happen without you. Join us at MIX09 to help pave the way to the Next Web now. Learn more and register at http://ad.doubleclick.net/clk;208669438;13503038;i?http://2009.visitmix.com/ _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
Hi Felipe,
It didn't help. No extra debug information came out It displays and goes into infinite loop "Setting pipeline to Paused..." Regards vinod On Fri, Dec 5, 2008 at 2:17 PM, Felipe Contreras <[hidden email]> wrote:
-- Vinod James ------------------------------------------------------------------------------ SF.Net email is Sponsored by MIX09, March 18-20, 2009 in Las Vegas, Nevada. The future of the web can't happen without you. Join us at MIX09 to help pave the way to the Next Web now. Learn more and register at http://ad.doubleclick.net/clk;208669438;13503038;i?http://2009.visitmix.com/ _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
It is strange, please try
gst-launch-0.10 audiotestsrc ! alsasink --gst-debug=alsa:5,basesink:4,baseaudiosink:4,audiosink:4 It is supposed to print a lot of log messages. If there is too much log, you can change all log level to 3 or remove some log. 2008/12/5 vinod james <[hidden email]> Hi Felipe, ------------------------------------------------------------------------------ SF.Net email is Sponsored by MIX09, March 18-20, 2009 in Las Vegas, Nevada. The future of the web can't happen without you. Join us at MIX09 to help pave the way to the Next Web now. Learn more and register at http://ad.doubleclick.net/clk;208669438;13503038;i?http://2009.visitmix.com/ _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
Hi,
This is the log it generates( I have attached as .txt with better formatting) 0:00:00.036590000 13368 0x8578098 DEBUG alsa gstalsasink.c:271:gst_alsasink_init:<GstAlsaSink@0x85c2170> initializing alsasink 0:00:00.036900000 13368 0x8578098 DEBUG alsa gstalsasink.c:301:gst_alsasink_getcaps:<alsasink0> device not open, using template caps Setting pipeline to PAUSED ... 0:00:00.037299000 13368 0x8578098 DEBUG audiosink gstaudiosink.c:565:gst_audio_sink_create_ringbuffer:<alsasink0> creating ringbuffer 0:00:00.037384000 13368 0x8578098 DEBUG audiosink gstaudiosink.c:567:gst_audio_sink_create_ringbuffer:<alsasink0> created ringbuffer @0x85c4908 0:00:00.051408000 13368 0x8578098 LOG alsa gstalsasink.c:676:gst_alsasink_open:<alsasink0> Opened device default 0:00:00.051607000 13368 0x8578098 DEBUG basesink gstbasesink.c:3456:gst_base_sink_change_state:<alsasink0> READY to PAUSED 0:00:00.051639000 13368 0x8578098 DEBUG basesink gstbasesink.c:3474:gst_base_sink_change_state:<alsasink0> doing async state change 0:00:00.051683000 13368 0x8578098 DEBUG basesink gstbasesink.c:2918:gst_base_sink_pad_activate:<alsasink0> Trying pull mode first 0:00:00.051710000 13368 0x8578098 DEBUG basesink gstbasesink.c:2927:gst_base_sink_pad_activate:<alsasink0> Falling back to push mode 0:00:00.051735000 13368 0x8578098 DEBUG basesink gstbasesink.c:2929:gst_base_sink_pad_activate:<alsasink0> Success activating push mode 0:00:00.052103000 13368 0x8578098 WARN alsa gstalsa.c:124:gst_alsa_detect_formats:<alsasink0> skipping non-int format 0:00:00.052144000 13368 0x8578098 LOG alsa gstalsa.c:30:gst_alsa_detect_rates:<alsasink0> probing sample rates ... 0:00:00.052171000 13368 0x8578098 DEBUG alsa gstalsa.c:49:gst_alsa_detect_rates:<alsasink0> Min. rate = 4000 (4000) 0:00:00.052194000 13368 0x8578098 DEBUG alsa gstalsa.c:50:gst_alsa_detect_rates:<alsasink0> Max. rate = 2147483647 (-1) 0:00:00.052219000 13368 0x8578098 LOG alsa gstalsa.c:265:gst_alsa_detect_channels:<alsasink0> probing channels ... 0:00:00.052240000 13368 0x8578098 DEBUG alsa gstalsa.c:309:gst_alsa_detect_channels:<alsasink0> Min. channels = 1 (1) 0:00:00.052262000 13368 0x8578098 DEBUG alsa gstalsa.c:310:gst_alsa_detect_channels:<alsasink0> Max. channels = 8 (10000) 0:00:00.052544000 13368 0x8578098 DEBUG alsa gstalsa.c:388:gst_alsa_open_iec958_pcm:<alsasink0> Generated device string "iec958:{AES0 0x02 AES1 0x82 AES2 0x00 AES3 0x02}" At this point I had to press Ctrl-d as it was in infinite loop Caught interrupt -- 0:00:05.107535000 13368 0x8578098 WARN alsa pcm_hw.c:1155:snd_pcm_hw_open: alsalib error: open /dev/snd/pcmC0D0p failed: Interrupted system call 0:00:05.107593000 13368 0x8578098 DEBUG alsa gstalsa.c:394:gst_alsa_open_iec958_pcm:<alsasink0> failed opening IEC958 device: Interrupted system call 0:00:05.107623000 13368 0x8578098 INFO alsa gstalsasink.c:321:gst_alsasink_getcaps:<alsasink0> returning caps audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false }, width=(int)32, depth=(int)32, rate=(int)[ 4000, 2147483647 ], channels=(int)[1, 2 ]; audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false}, width=(int)32, depth=(int)32, rate=(int)[ 4000, 2147483647 ],channels=(int)3, channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_LFE >; audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false }, width=(int)32, depth=(int)32, rate=(int)[4000, 2147483647 ], channels=(int)4,channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT >; audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false },width=(int)32, depth=(int)32, rate=(int)[ 4000, 2147483647 ], channels=(int)6, channel-positions=(GstAudioChannelPosition)<GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE >;audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false }, width=(int)32, depth=(int)32, rate=(int)[ 4000, 2147483647 ], channels=(int)8,channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE,GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT >; audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false },width=(int)24, depth=(int)24, rate=(int)[ 4000, 2147483647 ], channels=(int)[ 1, 2 ]; audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false}, width=(int)24, depth=(int)24, rate=(int)[ 4000, 2147483647 ], channels=(int)3, channel-positions=(GstAudioChannelPosition)<GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_LFE >; audio/x-raw-int, endianness=(int)1234,signed=(boolean){ true, false }, width=(int)24, depth=(int)24, rate=(int)[ 4000, 2147483647 ], channels=(int)4,channel-positions=(GstAudioChannelPosition)<GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT >;audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false }, width=(int)24, depth=(int)24, rate=(int)[ 4000, 2147483647 ], channels=(int)6,channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE >;audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false }, width=(int)24, depth=(int)24, rate=(int)[ 4000, 2147483647 ], channels=(int)8,channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE,GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT >; audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false },width=(int)32, depth=(int)24, rate=(int)[ 4000, 2147483647 ], channels=(int)[ 1, 2 ]; audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false}, width=(int)32, depth=(int)24, rate=(int)[ 4000, 2147483647 ], channels=(int)3, channel-positions=(GstAudioChannelPosition)<GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_LFE >; audio/x-raw-int, endianness=(int)1234,signed=(boolean){ true, false }, width=(int)32, depth=(int)24, rate=(int)[ 4000, 2147483647 ], channels=(int)4,channel-positions=(GstAudioChannelPosition)<GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT >;audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false }, width=(int)32, depth=(int)24, rate=(int)[ 4000, 2147483647 ], channels=(int)6,channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE >;audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false }, width=(int)32, depth=(int)24, rate=(int)[ 4000, 2147483647 ], channels=(int)8,channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE,GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT >; audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false },width=(int)16, depth=(int)16, rate=(int)[ 4000, 2147483647 ], channels=(int)[ 1, 2 ]; audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false}, width=(int)16, depth=(int)16, rate=(int)[ 4000, 2147483647 ], channels=(int)3, channel-positions=(GstAudioChannelPosition)<GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_LFE >; audio/x-raw-int, endianness=(int)1234,signed=(boolean){ true, false }, width=(int)16, depth=(int)16, rate=(int)[ 4000, 2147483647 ], channels=(int)4,channel-positions=(GstAudioChannelPosition)<GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT >;audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false }, width=(int)16, depth=(int)16, rate=(int)[ 4000, 2147483647 ], channels=(int)6,channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE >;audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false }, width=(int)16, depth=(int)16, rate=(int)[ 4000, 2147483647 ], channels=(int)8,channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE,GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT >; audio/x-raw-int, signed=(boolean){ true, false }, width=(int)8, depth=(int)8,rate=(int)[ 4000, 2147483647 ], channels=(int)[ 1, 2 ]; audio/x-raw-int, signed=(boolean){ true, false }, width=(int)8, depth=(int)8, rate=(int)[ 4000,2147483647 ], channels=(int)3, channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_LFE >; audio/x-raw-int, signed=(boolean){ true, false }, width=(int)8, depth=(int)8, rate=(int)[ 4000, 2147483647 ],channels=(int)4, channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT >; audio/x-raw-int, signed=(boolean){ true, false }, width=(int)8, depth=(int)8,rate=(int)[ 4000, 2147483647 ], channels=(int)6, channel-positions=(GstAudioChannelPosition)<GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE >;audio/x-raw-int, signed=(boolean){ true, false }, width=(int)8, depth=(int)8, rate=(int)[ 4000, 2147483647 ], channels=(int)8,channel-positions=(GstAudioChannelPosition)<GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE, GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT > Pipeline is PREROLLING ... handling interrupt. Interrupt: Stopping pipeline ... ERROR: pipeline doesn't want to preroll. Setting pipeline to NULL ... 0:00:05.194465000 13368 0x8578098 DEBUG basesink gstbasesink.c:2885:gst_base_sink_set_flushing:<alsasink0> flushing out data thread, need preroll to TRUE 0:00:05.194496000 13368 0x8578098 DEBUG basesink gstbasesink.c:1129:gst_base_sink_preroll_queue_flush:<alsasink0> flushing queue 0x85c2170 0:00:05.194529000 13368 0x8578098 DEBUG basesink gstbasesink.c:3608:gst_base_sink_change_state:<alsasink0> PAUSED to READY, posting async-done FREEING pipeline ...
On Fri, Dec 5, 2008 at 3:52 PM, Nie Jun <[hidden email]> wrote: It is strange, please try -- Vinod James ------------------------------------------------------------------------------ SF.Net email is Sponsored by MIX09, March 18-20, 2009 in Las Vegas, Nevada. The future of the web can't happen without you. Join us at MIX09 to help pave the way to the Next Web now. Learn more and register at http://ad.doubleclick.net/clk;208669438;13503038;i?http://2009.visitmix.com/ _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel log.txt (16K) Download Attachment |
Hi Felipe & Nie,
Any ideas on how to solve this problem? Regards vinod On Fri, Dec 5, 2008 at 4:49 PM, vinod james <[hidden email]> wrote: Hi, -- Vinod James ------------------------------------------------------------------------------ SF.Net email is Sponsored by MIX09, March 18-20, 2009 in Las Vegas, Nevada. The future of the web can't happen without you. Join us at MIX09 to help pave the way to the Next Web now. Learn more and register at http://ad.doubleclick.net/clk;208669438;13503038;i?http://2009.visitmix.com/ _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
vinod james schrieb:
> Hi Felipe & Nie, > Any ideas on how to solve this problem? > Regards > vinod > > On Fri, Dec 5, 2008 at 4:49 PM, vinod james <[hidden email] > <mailto:[hidden email]>> wrote: > > Hi, > This is the log it generates( I have attached as .txt with better > formatting) > 0:00:00.036590000 13368 0x8578098 DEBUG alsa > gstalsasink.c:271:gst_alsasink_init:<GstAlsaSink@0x85c2170> > initializing alsasink > 0:00:00.036900000 13368 0x8578098 DEBUG alsa > gstalsasink.c:301:gst_alsasink_getcaps:<alsasink0> device not open, > using template caps > > Setting pipeline to PAUSED ... > > ... > At this point I had to press Ctrl-d as it was in infinite loop better redirect to a file and sent it for analysis. GST_DEBUG=*:2,alsa:5 GST_DEBUG_NO_COLOR=1 gst-launch-0.10 2>debug.log audiotestsrc num-buffers=10 ! alsasink Stefan ------------------------------------------------------------------------------ SF.Net email is Sponsored by MIX09, March 18-20, 2009 in Las Vegas, Nevada. The future of the web can't happen without you. Join us at MIX09 to help pave the way to the Next Web now. Learn more and register at http://ad.doubleclick.net/clk;208669438;13503038;i?http://2009.visitmix.com/ _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
Just a thought, I've had issues with alsasink caps which I've resolved
by using alsa's plughw interface: GST_DEBUG=*:2,alsa:5 GST_DEBUG_NO_COLOR=1 gst-launch-0.10 2>debug.log audiotestsrc num-buffers=10 ! alsasink device=plughw:0 I don't know if this will resolve your issue but I'd be curious to see the debug output. -Tristan Stefan Kost wrote: > vinod james schrieb: > >> Hi Felipe & Nie, >> Any ideas on how to solve this problem? >> Regards >> vinod >> >> On Fri, Dec 5, 2008 at 4:49 PM, vinod james <[hidden email] >> <mailto:[hidden email]>> wrote: >> >> Hi, >> This is the log it generates( I have attached as .txt with better >> formatting) >> 0:00:00.036590000 13368 0x8578098 DEBUG alsa >> gstalsasink.c:271:gst_alsasink_init:<GstAlsaSink@0x85c2170> >> initializing alsasink >> 0:00:00.036900000 13368 0x8578098 DEBUG alsa >> gstalsasink.c:301:gst_alsasink_getcaps:<alsasink0> device not open, >> using template caps >> >> Setting pipeline to PAUSED ... >> >> ... >> At this point I had to press Ctrl-d as it was in infinite loop >> > > better redirect to a file and sent it for analysis. > GST_DEBUG=*:2,alsa:5 GST_DEBUG_NO_COLOR=1 gst-launch-0.10 2>debug.log > audiotestsrc num-buffers=10 ! alsasink > > > Stefan > > ------------------------------------------------------------------------------ > SF.Net email is Sponsored by MIX09, March 18-20, 2009 in Las Vegas, Nevada. > The future of the web can't happen without you. Join us at MIX09 to help > pave the way to the Next Web now. Learn more and register at > http://ad.doubleclick.net/clk;208669438;13503038;i?http://2009.visitmix.com/ > _______________________________________________ > gstreamer-devel mailing list > [hidden email] > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- Tristan Matthews Société des arts technologiques [SAT] email: [hidden email] web: http://www.music.mcgill.ca/~tmatthews ------------------------------------------------------------------------------ SF.Net email is Sponsored by MIX09, March 18-20, 2009 in Las Vegas, Nevada. The future of the web can't happen without you. Join us at MIX09 to help pave the way to the Next Web now. Learn more and register at http://ad.doubleclick.net/clk;208669438;13503038;i?http://2009.visitmix.com/ _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
In reply to this post by vinod james
Hi, gstreamer-devel:
I think you'd better add some "g_message" or "printf" codes into your gstbasesink.c and find out where the pipeline blocked, that's the key point. Blocking is a common problem when developing gstreamer applications, at least for me. And my solution is first find out where it blocks -- mostly always blocks at basesink. I noticed your basesink used push mode so you can first add printf codes in "gst_base_sink_chain" --> "gst_base_sink_chain_unlock" --> "gst_base_sink_queue_object_unlocked" ...... Eric Zhang 2008/12/5 vinod james <[hidden email]>
------------------------------------------------------------------------------ SF.Net email is Sponsored by MIX09, March 18-20, 2009 in Las Vegas, Nevada. The future of the web can't happen without you. Join us at MIX09 to help pave the way to the Next Web now. Learn more and register at http://ad.doubleclick.net/clk;208669438;13503038;i?http://2009.visitmix.com/ _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
Hi, gstreamer-devel:
One more thing: alsasink inherits from gstbaseaudiosink which inherits from gstbasesink. GstBaseAudioSink overrides "render" function of GstBaseSink, so don't forget to take care of gstbaseaudiosink. Eric Zhang 2008/12/9 Eric Zhang <[hidden email]> Hi, gstreamer-devel: ------------------------------------------------------------------------------ SF.Net email is Sponsored by MIX09, March 18-20, 2009 in Las Vegas, Nevada. The future of the web can't happen without you. Join us at MIX09 to help pave the way to the Next Web now. Learn more and register at http://ad.doubleclick.net/clk;208669438;13503038;i?http://2009.visitmix.com/ _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
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